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SIP/2.0 400 Invalid Contact information

Question
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Hello,
I have a problem with incoming SIP calls. Here is my test environement...
MOC2007 <-> OCS2007 <-> Mediation <-> VoIP Gateway
Calls outbound from my MOC2007 client work perfectly, however incoming calls from my VoIP Gateway get returned "SIP/2.0 400 Invalid Contact information". It is actually the OCS2007 server returning this message after receiving the forwarded INVITE from the Mediation server.
Here is the trace.
TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-08:10:41.866.000001c4 (S4,SipMessage.DataLoggingHelper:472.idx(550))
<<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_3D8BD1E>], 192.168.195.114:5060<-192.168.195.190:1150
INVITE sip:+33497234852@192.168.195.114:5060;transport=tcp SIP/2.0
FROM: <sip:0630704615@192.168.195.114>;tag=q-7033-bae9
TO: <sip:+33497234852@192.168.195.114>
CSEQ: 4 INVITE
CALL-ID: 12840797339373401@192.168.195.190
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKlwmphvnC00658927393
RECORD-ROUTE: <sip:192.168.195.190>
CONTACT: <sip:0630704615@192.168.195.190;transport=tcp>
CONTENT-LENGTH: 251
USER-AGENT: QuesCom SIP Gateway 5.00.007
CONTENT-TYPE: application/sdp
ALLOW: INVITE, BYE, ACK, CANCEL, REGISTER, OPTIONS, REFER, NOTIFY, INFO
v=0
o=QuesCom 4308 4308 IN IP4 192.168.195.190
s=NonSIP
c=IN IP4 192.168.195.190
t=0 0
m=audio 11036 RTP/AVP 8 0 18 101
a=rtpmap:8 pcma/8000/1
a=rtpmap:0 pcmu/8000/1
a=rtpmap:18 g729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
------------EndOfIncoming SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-08:10:41.866.000001d0 (S4,SipMessage.DataLoggingHelper:472.idx(500))
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_3D8BD1E>], 192.168.195.114:5060->192.168.195.190:1150
SIP/2.0 100 Trying
FROM: <sip:0630704615@192.168.195.114>;tag=q-7033-bae9
TO: <sip:+33497234852@192.168.195.114>
CSEQ: 4 INVITE
CALL-ID: 12840797339373401@192.168.195.190
VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKlwmphvnC00658927393
CONTENT-LENGTH: 0
------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-08:10:41.946.0000022e (S4,SipMessage.DataLoggingHelper:472.idx(500))
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_3571E9F>], 192.168.195.114:2883->192.168.195.113:5061
INVITE sip:+33497234852@testlcs.local;user=phone SIP/2.0
FROM: <sip:0630704615;phone-context=unknown@testlcs.local;user=phone>;epid=50FADD4DA6;tag=45f6d9ea4
TO: <sip:+33497234852@testlcs.local;user=phone>
CSEQ: 11 INVITE
CALL-ID: bdf869a2-bca4-4ff7-b7d0-022d3709c9ff
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 192.168.195.114:2883;branch=z9hG4bK4a8ec5a0
CONTACT: <sipcottlcs2007med.testlcs.local@testlcs.local;gruu;opaque=srvr:MediationServer:mc3q_s6qrk-5EezN3UoChgAA;grid=d12c4066519c4f8baa1446bac4af551c>;isGateway
CONTENT-LENGTH: 944
SUPPORTED: replaces
SUPPORTED: gruu-10
USER-AGENT: RTCC/3.0.0.0 MediationServer
CONTENT-TYPE: application/sdp; charset=utf-8
ALLOW: UPDATE
ms-call-source: non-ms-rtc
ALLOW: Ack, Cancel, Bye,Invite,Refer
v=0
o=- 0 0 IN IP4 192.168.195.114
s=session
c=IN IP4 192.168.195.114
b=CT:1000
t=0 0
m=audio 62053 RTP/AVP 97 101 115 111 0 8
c=IN IP4 192.168.195.114
a=rtcp:63607
a=candidate:qAIY9y4X7Bw0HEpXWZ98oqsjTz6HlnZC44YyNO/bg+o 1 4QhvRWT/npPPNmNIIUhesw UDP 0.900 192.168.195.114 62053
a=candidate:qAIY9y4X7Bw0HEpXWZ98oqsjTz6HlnZC44YyNO/bg+o 2 4QhvRWT/npPPNmNIIUhesw UDP 0.900 192.168.195.114 63607
a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:9kOZ+aKWXlO+Mc5ETULXMUTxW8efQ1j7iiNVlUCC|2^31|1:1
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:+QL1qybw9F8+wLv7WGLNJAB5R77IyT3rWnZPzpr9|2^31|1:1
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:7bEKYRZNsIL4Sd0TqnPz98eHgTLVwETXK2iqnLjs|2^31
a=label:main-audio
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:115 x-msrta/8000
a=fmtp:115 bitrate=11800
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-08:10:43.799.00000253 (S4,SipMessage.DataLoggingHelper:472.idx(550))
<<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_3571E9F>], 192.168.195.114:2883<-192.168.195.113:5061
SIP/2.0 400 Invalid Contact information
FROM: <sip:0630704615;phone-context=unknown@testlcs.local;user=phone>;epid=50FADD4DA6;tag=45f6d9ea4
TO: <sip:+33497234852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
CSEQ: 11 INVITE
CALL-ID: bdf869a2-bca4-4ff7-b7d0-022d3709c9ff
VIA: SIP/2.0/TLS 192.168.195.114:2883;branch=z9hG4bK4a8ec5a0;ms-received-port=2883;ms-received-cid=29200
CONTENT-LENGTH: 0
ms-diagnostics: 1018;reason="Parsing failure";source="scottlcs2007.testlcs.local"
------------EndOfIncoming SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-08:10:43.799.00000262 (S4,SipMessage.DataLoggingHelper:472.idx(500))
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_3571E9F>], 192.168.195.114:2883->192.168.195.113:5061
ACK sip:+33497234852@testlcs.local;user=phone SIP/2.0
FROM: <sip:0630704615;phone-context=unknown@testlcs.local;user=phone>;tag=45f6d9ea4;epid=50FADD4DA6
TO: <sip:+33497234852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
CSEQ: 11 ACK
CALL-ID: bdf869a2-bca4-4ff7-b7d0-022d3709c9ff
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 192.168.195.114:2883;branch=z9hG4bK4a8ec5a0
CONTENT-LENGTH: 0
------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-08:10:43.889.000002c5 (S4,SipMessage.DataLoggingHelper:472.idx(500))
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_208E51>], 192.168.195.114:2884->192.168.195.113:5061
SERVICE sip:+33497234852@testlcs.local;user=phone SIP/2.0
FROM: <sip:+33497234852@testlcs.local;user=phone>;epid=50FADD4DA6;tag=ca38349310
TO: <sip:+33497234852@testlcs.local;user=phone>
CSEQ: 12 SERVICE
CALL-ID: 36a1168028fd45da882d2b5af58c2ecd
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 192.168.195.114:2884;branch=z9hG4bK4e403b9a
CONTENT-LENGTH: 500
USER-AGENT: RTCC/3.0.0.0 MediationServer
CONTENT-TYPE: application/msrtc-reporterror+xml
<?xml version="1.0" encoding="us-ascii"?><reportError xmlns="http://schemas.microsoft.com/2006/09/sip/error-reporting"><error callId="bdf869a2-bca4-4ff7-b7d0-022d3709c9ff" toUri="sip:+33497234852@testlcs.local;user=phone" fromTag="45f6d9ea4" toTag="FE355C133B228334B697EF1833317D62" requestType="INVITE" contentType="application/sdp;call-type=audio" responseCode="400"><diagHeader>1018;reason="Parsing failure";source="scottlcs2007.testlcs.local"</diagHeader><progressReports /></error></reportError>------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-08:10:43.889.000002cb (S4,SipMessage.DataLoggingHelper:472.idx(500))
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_3D8BD1E>], 192.168.195.114:5060->192.168.195.190:1150
SIP/2.0 400 Invalid Contact information
FROM: <sip:0630704615@192.168.195.114>;tag=q-7033-bae9
TO: <sip:+33497234852@192.168.195.114>;tag=b82053edfc
CSEQ: 4 INVITE
CALL-ID: 12840797339373401@192.168.195.190
VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKlwmphvnC00658927393
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0 MediationServer
------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-08:10:44.109.00000322 (S4,SipMessage.DataLoggingHelper:472.idx(550))
<<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_208E51>], 192.168.195.114:2884<-192.168.195.113:5061
SIP/2.0 401 Unauthorized
FROM: <sip:+33497234852@testlcs.local;user=phone>;epid=50FADD4DA6;tag=ca38349310
TO: <sip:+33497234852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
CSEQ: 12 SERVICE
CALL-ID: 36a1168028fd45da882d2b5af58c2ecd
VIA: SIP/2.0/TLS 192.168.195.114:2884;branch=z9hG4bK4e403b9a;ms-received-port=2884;ms-received-cid=29300
WWW-AUTHENTICATE: NTLM realm="SIP Communications Service", targetname="scottlcs2007.testlcs.local", version=3
WWW-AUTHENTICATE: Kerberos realm="SIP Communications Service", targetname="sip/scottlcs2007.testlcs.local", version=3
CONTENT-LENGTH: 0
DATE: Thu, 29 Nov 2007 08:10:38 GMT
------------EndOfIncoming SipMessageWhy is the OCS2007 server returning this message ?
How do I correct this ?
Thanks for your help.
Sott
Thursday, November 29, 2007 9:23 AM
All replies
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Hi Scott,
Try to have the media gateway present the From number as E.164 with a + prefixed to the Mediation Server. Also make sure you have configured a location profile on the Mediation Server and related normalization rules and have configured your user correctly with msRTCSIP-Line. Have a look in the OCS_VoIP_Guide.
Also I don't think your media gateway is on the list of qualified media gateways http://technet.microsoft.com/en-us/office/bb735838.aspx, so you might run into more problems.
best regards
Jens
Thursday, November 29, 2007 10:34 AM -
I believe I have set up the system correctly, but it still does not work. I now send the + in the FROM field.
TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-11:50:12.430.00000b65 (S4,SipMessage.DataLoggingHelper:472.idx(550))
<<<<<<<<<<<<Incoming SipMessage c=[<SipTcpConnection_8B4A0E>], 192.168.195.114:5060<-192.168.195.190:1087
INVITE sip:+4852@192.168.195.114:5060;transport=tcp SIP/2.0
FROM: <sip:+33630704615@192.168.195.114>;tag=q-18be-ee5d
TO: <sip:+4852@192.168.195.114>
CSEQ: 1 INVITE
CALL-ID: 12840810511191070@192.168.195.190
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKfrbfjbdC001543211
RECORD-ROUTE: <sip:192.168.195.190>
CONTACT: <sip:+33630704615@192.168.195.190;transport=tcp>
CONTENT-LENGTH: 253
USER-AGENT: QuesCom SIP Gateway 5.00.007
CONTENT-TYPE: application/sdp
ALLOW: INVITE, BYE, ACK, CANCEL, REGISTER, OPTIONS, REFER, NOTIFY, INFO
v=0
o=QuesCom 18467 18467 IN IP4 192.168.195.190
s=NonSIP
c=IN IP4 192.168.195.190
t=0 0
m=audio 11010 RTP/AVP 8 0 18 101
a=rtpmap:8 pcma/8000/1
a=rtpmap:0 pcmu/8000/1
a=rtpmap:18 g729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
------------EndOfIncoming SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:12.440.00000b71 (S4,SipMessage.DataLoggingHelper:472.idx(500))
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_8B4A0E>], 192.168.195.114:5060->192.168.195.190:1087
SIP/2.0 100 Trying
FROM: <sip:+33630704615@192.168.195.114>;tag=q-18be-ee5d
TO: <sip:+4852@192.168.195.114>
CSEQ: 1 INVITE
CALL-ID: 12840810511191070@192.168.195.190
VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKfrbfjbdC001543211
CONTENT-LENGTH: 0
------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:12.530.00000bff (S4,SipMessage.DataLoggingHelper:472.idx(500))
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_3300B4D>], 192.168.195.114:3684->192.168.195.113:5061
INVITE sip:+4852@testlcs.local;user=phone SIP/2.0
FROM: <sip:+33630704615@testlcs.local;user=phone>;epid=50FADD4DA6;tag=e8952cc14
TO: <sip:+4852@testlcs.local;user=phone>
CSEQ: 21 INVITE
CALL-ID: 5e80aad3-8fca-4f90-bc1b-db6a9baf35bf
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 192.168.195.114:3684;branch=z9hG4bK571d6d76
CONTACT: <sipcottlcs2007med.testlcs.local@testlcs.local;gruu;opaque=srvr:MediationServer:mc3q_s6qrk-5EezN3UoChgAA;grid=c60f17471e554939854cdcd79523abfc>;isGateway
CONTENT-LENGTH: 944
SUPPORTED: replaces
SUPPORTED: gruu-10
USER-AGENT: RTCC/3.0.0.0 MediationServer
CONTENT-TYPE: application/sdp; charset=utf-8
ALLOW: UPDATE
ms-call-source: non-ms-rtc
ALLOW: Ack, Cancel, Bye,Invite,Refer
v=0
o=- 0 0 IN IP4 192.168.195.114
s=session
c=IN IP4 192.168.195.114
b=CT:1000
t=0 0
m=audio 61355 RTP/AVP 97 101 115 111 0 8
c=IN IP4 192.168.195.114
a=rtcp:62123
a=candidatevSSEml9F2cnJP93SGZ8bu7msFdcHmm80vSLWZTdVLM 1 omXUt93kRdTdCzHCq9D+9A UDP 0.900 192.168.195.114 61355
a=candidatevSSEml9F2cnJP93SGZ8bu7msFdcHmm80vSLWZTdVLM 2 omXUt93kRdTdCzHCq9D+9A UDP 0.900 192.168.195.114 62123
a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:jJPJeA1RC89pyUK/XPtcyytVP84gRDi5rCgQTq7K|2^31|1:1
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:cPvwZ5OF9N9zutdg2JoQIV6XtFpuYULob027AB7w|2^31|1:1
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:QjWjKn1QvQIPfhj+ofhp93006LYhIsb6sazBwwjy|2^31
a=label:main-audio
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:115 x-msrta/8000
a=fmtp:115 bitrate=11800
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-11:50:14.393.00000c24 (S4,SipMessage.DataLoggingHelper:472.idx(550))
<<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_3300B4D>], 192.168.195.114:3684<-192.168.195.113:5061
SIP/2.0 400 Invalid Contact information
FROM: <sip:+33630704615@testlcs.local;user=phone>;epid=50FADD4DA6;tag=e8952cc14
TO: <sip:+4852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
CSEQ: 21 INVITE
CALL-ID: 5e80aad3-8fca-4f90-bc1b-db6a9baf35bf
VIA: SIP/2.0/TLS 192.168.195.114:3684;branch=z9hG4bK571d6d76;ms-received-port=3684;ms-received-cid=29E00
CONTENT-LENGTH: 0
ms-diagnostics: 1018;reason="Parsing failure";source="scottlcs2007.testlcs.local"
------------EndOfIncoming SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:14.393.00000c33 (S4,SipMessage.DataLoggingHelper:472.idx(500))
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_3300B4D>], 192.168.195.114:3684->192.168.195.113:5061
ACK sip:+4852@testlcs.local;user=phone SIP/2.0
FROM: <sip:+33630704615@testlcs.local;user=phone>;tag=e8952cc14;epid=50FADD4DA6
TO: <sip:+4852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
CSEQ: 21 ACK
CALL-ID: 5e80aad3-8fca-4f90-bc1b-db6a9baf35bf
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 192.168.195.114:3684;branch=z9hG4bK571d6d76
CONTENT-LENGTH: 0
------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:14.503.00000c5b (S4,SipMessage.DataLoggingHelper:472.idx(500))
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTcpConnection_8B4A0E>], 192.168.195.114:5060->192.168.195.190:1087
SIP/2.0 400 Invalid Contact information
FROM: <sip:+33630704615@192.168.195.114>;tag=q-18be-ee5d
TO: <sip:+4852@192.168.195.114>;tag=1ead26a45
CSEQ: 1 INVITE
CALL-ID: 12840810511191070@192.168.195.190
VIA: SIP/2.0/TCP 192.168.195.190:5060;branch=z9hG4bKfrbfjbdC001543211
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0 MediationServer
------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.08C8::11/29/2007-11:50:14.513.00000cca (S4,SipMessage.DataLoggingHelper:472.idx(500))
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_1B76E14>], 192.168.195.114:3685->192.168.195.113:5061
SERVICE sip:+4852@testlcs.local;user=phone SIP/2.0
FROM: <sip:+4852@testlcs.local;user=phone>;epid=50FADD4DA6;tag=89af6120c0
TO: <sip:+4852@testlcs.local;user=phone>
CSEQ: 22 SERVICE
CALL-ID: 9454d7a1041648d0a2f0f229299ff1f2
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 192.168.195.114:3685;branch=z9hG4bKf4b48eb2
CONTENT-LENGTH: 493
USER-AGENT: RTCC/3.0.0.0 MediationServer
CONTENT-TYPE: application/msrtc-reporterror+xml
<?xml version="1.0" encoding="us-ascii"?><reportError xmlns="http://schemas.microsoft.com/2006/09/sip/error-reporting"><error callId="5e80aad3-8fca-4f90-bc1b-db6a9baf35bf" toUri="sip:+4852@testlcs.local;user=phone" fromTag="e8952cc14" toTag="FE355C133B228334B697EF1833317D62" requestType="INVITE" contentType="application/sdp;call-type=audio" responseCode="400"><diagHeader>1018;reason="Parsing failure";source="scottlcs2007.testlcs.local"</diagHeader><progressReports /></error></reportError>------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [0]09C4.0ACC::11/29/2007-11:50:16.796.00000cf3 (S4,SipMessage.DataLoggingHelper:472.idx(550))
<<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_1B76E14>], 192.168.195.114:3685<-192.168.195.113:5061
SIP/2.0 401 Unauthorized
FROM: <sip:+4852@testlcs.local;user=phone>;epid=50FADD4DA6;tag=89af6120c0
TO: <sip:+4852@testlcs.local;user=phone>;tag=FE355C133B228334B697EF1833317D62
CSEQ: 22 SERVICE
CALL-ID: 9454d7a1041648d0a2f0f229299ff1f2
VIA: SIP/2.0/TLS 192.168.195.114:3685;branch=z9hG4bKf4b48eb2;ms-received-port=3685;ms-received-cid=29F00
WWW-AUTHENTICATE: NTLM realm="SIP Communications Service", targetname="scottlcs2007.testlcs.local", version=3
WWW-AUTHENTICATE: Kerberos realm="SIP Communications Service", targetname="sip/scottlcs2007.testlcs.local", version=3
CONTENT-LENGTH: 0
DATE: Thu, 29 Nov 2007 11:50:14 GMT
------------EndOfIncoming SipMessageThanks again for your help.
Scott
Thursday, November 29, 2007 1:25 PM -
Same problem on my OCS <-> VOIP Gateway settlement. Calling from Communicator to Gateway users run fine; every call originated by gateway users receive SIP/2.0 400 Invalid Contact Information. Which is the Normalization rule role in inbound OCS calls ? I have a location profile normalization rule: ^\+(\d{5})$ <--> $1 for manage outbond calls, it's checked for inbound calls too ?
This is the SIP log: (Every attempt with (+) in INVITE/FROM/TO header was done. Same result)
INVITE sip:44653@mediation.tsf.local;transport=tcp SIP/2.0
Record-Route: <sip:10.57.3.141:5080;lr;sipX-route=%2Afrom%7EYzE0N2RiMWI%60.authrules%2Aauth%7E%21822274c8a0ce72388ed7b8363600c832>
Max-Forwards: 18
Contact: <sip:+200@10.57.7.20:17216>
To: "+44653"<sip:+44653@10.57.3.141>
From: "+200"<sip:+200@10.57.3.141>;tag=c147db1b
Call-Id: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.
Cseq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 412
Date: Wed, 09 Jan 2008 10:26:18 GMT
Via: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b
Via: SIP/2.0/UDP 10.57.3.141;branch=z9hG4bK-c737ffd3a2dba9aec9a47ba877f2603c
Via: SIP/2.0/UDP 10.57.7.20:17216;branch=z9hG4bK-d87543-ad437625ba4cde23-1--d87543-;rport=17216
v=0
o=- 2 2 IN IP4 10.57.7.20
s=CounterPath X-Lite 3.0
c=IN IP4 10.57.7.20
t=0 0
m=audio 5006 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 1 : UTFJi2Ky +xViCroo 10.57.7.20 5006
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
SIP/2.0 100 Trying
FROM: "+200"<sip:+200@10.57.3.141>;tag=c147db1b
TO: "+44653"<sip:+44653@10.57.3.141>
CSEQ: 1 INVITE
CALL-ID: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.
VIA: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b,SIP/2.0/UDP 10.57.3.141;branch=z9hG4bK-c737ffd3a2dba9aec9a47ba877f2603c,SIP/2.0/UDP 10.57.7.20:17216;branch=z9hG4bK-d87543-ad437625ba4cde23-1--d87543-;rport=17216
CONTENT-LENGTH: 0
SIP/2.0 400 Invalid Contact information
FROM: "+200"<sip:+200@10.57.3.141>;tag=c147db1b
TO: "+44653"<sip:+44653@10.57.3.141>;tag=34475251be
CSEQ: 1 INVITE
CALL-ID: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.
VIA: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b,SIP/2.0/UDP 10.57.3.141;branch=z9hG4bK-c737ffd3a2dba9aec9a47ba877f2603c,SIP/2.0/UDP 10.57.7.20:17216;branch=z9hG4bK-d87543-ad437625ba4cde23-1--d87543-;rport=17216
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0 MediationServer
ACK sip:44653@mediation.tsf.local;transport=tcp SIP/2.0
Contact: sip:+200@10.57.7.20:17216
From: "+200"<sip:+200@10.57.3.141>;tag=c147db1b
To: "+44653"<sip:+44653@10.57.3.141>;tag=34475251be
Call-Id: ZjIyODIxOTRjOTM1YjkzMDI3OTllZjAzYjc2Y2ZmMWY.
Cseq: 1 ACK
Max-Forwards: 20
Via: SIP/2.0/TCP 10.57.3.141:5080;branch=z9hG4bK-a6658b09d29177b46b8edabfd6baf26b
Content-Length: 0Wednesday, January 9, 2008 11:10 AM -
Hi Alberto, has this issue been resolved? I'm experiencing exactly the same issue (posted yesterday) where I can't call from my Mitel IP phone to the Office Communicator client. Had no responses yet, so was wondering if your problem is already solved.
Thanks
Friday, January 18, 2008 11:20 AM -
Issue is still open for me. Maybe SIP/TCP stack of our VOIP Gateway is buggy/lack but I'm not be able to make calls from Dialogic Gateway to OCS mediation server work too.Monday, January 21, 2008 3:28 PM